Commit 99d61d34 authored by Diego Biurrun's avatar Diego Biurrun

cosmetics: prettyprinting, K&R style, break overly long lines

Originally committed as revision 19377 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent fd257dc4
This diff is collapsed.
......@@ -181,14 +181,14 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
ff_sine_window_init(ff_sine_1024, 1024);
ff_sine_window_init(ff_sine_128, 128);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
avctx->extradata = av_malloc(2);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
sizes[0] = swb_size_1024[i];
sizes[1] = swb_size_128[i];
sizes[0] = swb_size_1024[i];
sizes[1] = swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
......@@ -353,8 +353,10 @@ static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, in
for (w = 0; w < ics0->num_windows*16; w += 16)
for (i = 0; i < ics0->max_sfb; i++)
if (cpe->ms_mask[w+i]) msc++;
if (msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0;
else cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
if (msc == 0 || ics0->max_sfb == 0)
cpe->ms_mode = 0;
else
cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
}
}
......@@ -373,7 +375,8 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
/**
* Encode scalefactors.
*/
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce)
{
int off = sce->sf_idx[0], diff;
int i, w;
......@@ -382,7 +385,8 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, Single
for (i = 0; i < sce->ics.max_sfb; i++) {
if (!sce->zeroes[w*16 + i]) {
diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
if (diff < 0 || diff > 120) av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
if (diff < 0 || diff > 120)
av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
off = sce->sf_idx[w*16 + i];
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
......@@ -398,7 +402,8 @@ static void encode_pulses(AACEncContext *s, Pulse *pulse)
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if (!pulse->num_pulse) return;
if (!pulse->num_pulse)
return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
......@@ -424,10 +429,10 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
}
for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda);
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda);
}
start += sce->ics.swb_sizes[i];
}
......@@ -437,10 +442,13 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
if (!common_window) put_ics_info(s, &sce->ics);
if (!common_window)
put_ics_info(s, &sce->ics);
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
......@@ -453,7 +461,8 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, Si
/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
const char *name)
{
int i, namelen, padbits;
......@@ -484,20 +493,23 @@ static int aac_encode_frame(AVCodecContext *avctx,
return 0;
if (data) {
if (!s->psypp) {
memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
memcpy(s->samples + 1024 * avctx->channels, data,
1024 * avctx->channels * sizeof(s->samples[0]));
} else {
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for (i = 0; i < chan_map[0]; i++) {
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch, chans);
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
}
}
if (!avctx->frame_number) {
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return 0;
}
......@@ -509,11 +521,11 @@ static int aac_encode_frame(AVCodecContext *avctx,
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < chan_map[0]; i++) {
FFPsyWindowInfo wi[2];
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
if (!data) la = NULL;
for (j = 0; j < chans; j++) {
IndividualChannelStream *ics = &cpe->ch[j].ics;
......@@ -577,12 +589,14 @@ static int aac_encode_frame(AVCodecContext *avctx,
}
if (avctx->frame_bits > 6144*avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n", avctx->frame_bits, 6144*avctx->channels);
av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n",
avctx->frame_bits, 6144*avctx->channels);
}
if (!data)
s->last_frame = 1;
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
memcpy(s->samples, s->samples + 1024 * avctx->channels,
1024 * avctx->channels * sizeof(s->samples[0]));
return put_bits_count(&s->pb)>>3;
}
......
......@@ -106,7 +106,7 @@ static av_cold float calc_bark(float f)
static av_cold float ath(float f, float add)
{
f /= 1000.0f;
return 3.64 * pow(f, -0.8)
return 3.64 * pow(f, -0.8)
- 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
......@@ -181,11 +181,11 @@ static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
int channel, int prev_type)
{
int i, j;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
FFPsyWindowInfo wi;
memset(&wi, 0, sizeof(wi));
......@@ -199,12 +199,12 @@ static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
sum += v*v;
}
s[i] = sum;
s[i] = sum;
sum2 += sum;
}
for (i = 0; i < 8; i++) {
if (s[i] > pch->win_energy * attack_ratio) {
attack_n = i + 1;
attack_n = i + 1;
switch_to_eight = 1;
break;
}
......@@ -255,16 +255,16 @@ static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
/**
* Calculate band thresholds as suggested in 3GPP TS26.403
*/
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
FFPsyWindowInfo *wi)
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
const float *coefs, FFPsyWindowInfo *wi)
{
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
Psy3gppChannel *pch = &pctx->ch[channel];
int start = 0;
int i, w, g;
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
for (w = 0; w < wi->num_windows*16; w += 16) {
......@@ -274,8 +274,8 @@ static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
for (i = 0; i < band_sizes[g]; i++)
band->energy += coefs[start+i] * coefs[start+i];
band->energy *= 1.0f / (512*512);
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
}
......
......@@ -81,7 +81,7 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
FFPsyPreprocessContext *ctx;
int i;
float cutoff_coeff;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
ctx->avctx = avctx;
if (avctx->flags & CODEC_FLAG_QSCALE)
......@@ -90,7 +90,7 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
FILT_ORDER, cutoff_coeff, 0.0, 0.0);
FILT_ORDER, cutoff_coeff, 0.0, 0.0);
if (ctx->fcoeffs) {
ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for (i = 0; i < avctx->channels; i++)
......
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